The PhoneBoy Blog

Gizmo now supports registering against other SIP proxies!

23 May 2006

Michael Robertson announced the arrival of Gizmo Project 2.0. The main feature in this release is the ability to register your Gizmo client against Asterisk, but it should work with other SIP proxies as well.

Gizmo Project 2.0 now has support for Asterisk, making it possible to easily configure it to remotely receive and dial calls from from an Asterisk system on any net-connected PC. It’s nothing new to use software to accept Asterisk calls. What IS new is to be able to do it reliably. Oftentimes, Asterisk calls get blocked by routers, NATs and firewalls using other SIP software, so the phone rings but one or both parties can’t hear each other.

How do they do this reliably? It looks pretty simple to me–they proxy everything through the SIPPhone proxies. From a NAT traversal point of view, this is a good thing. Presumably they have already licked NAT traversal between a Gizmo client and the SIPPhone proxy servers. By registering the alternate SIP credentials from the SIPPhone proxy, which is directly on the Internet, they eliminate the potential for NAT traversal issues for the secondary SIP connection. I have some some sniffing and can ve that both the call control the media is proxied through SIPPhone’s servers.

From Asterisk’s point of view, it looks like this:

asterisk*CLI> sip show peer myext
asterisk*CLI> 

  * Name       : myext
  Secret       : <Set>
  MD5Secret    : <Not set>
  Context      : international-usa
  Subscr.Cont. : <Not set>
  Language     : 
  AMA flags    : Unknown
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    : 
  Pickupgroup  : 
  Mailbox      : myext@default
  VM Extension : asterisk
  LastMsgsSent : 2
  Call limit   : 0
  Dynamic      : Yes
  Callerid     : "" <>
  Expire       : 15370981
  Insecure     : no
  Nat          : Always
  ACL          : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  Trust RPID   : No
  Send RPID    : No
  DTMFmode     : rfc2833
  LastMsg      : 0
  ToHost       : 
  Addr->IP     : 198.65.166.131 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Def. Username: 1747xxxxxxx
  SIP Options  : (none)
  Codecs       : 0x50e (gsm|ulaw|alaw|g729|ilbc)
  Codec Order  : (ilbc,ulaw,alaw,gsm,g729)
  Status       : Unmonitored
  Useragent    : MacGizmo (Gizmo-s2n1)/1.5
  Reg. Contact : sip:[email protected]:5060

I have to give Michael Robertson and the rest of the crew over at Gizmo major kudos, this is a great feature!